/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 3 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
**********/
// "liveMedia"
// Copyright (c) 1996-2018 Live Networks, Inc.  All rights reserved.
// RTP sink for MPEG-4 audio, using LATM multiplexing (RFC 3016)
// Implementation

#include "include/MPEG4LATMAudioRTPSink.hh"

MPEG4LATMAudioRTPSink
::MPEG4LATMAudioRTPSink(UsageEnvironment &env, Groupsock *RTPgs,
                        u_int8_t rtpPayloadFormat,
                        u_int32_t rtpTimestampFrequency,
                        char const *streamMuxConfigString,
                        unsigned numChannels,
                        Boolean allowMultipleFramesPerPacket)
        : AudioRTPSink(env, RTPgs, rtpPayloadFormat,
                       rtpTimestampFrequency, "MP4A-LATM", numChannels),
          fStreamMuxConfigString(strDup(streamMuxConfigString)),
          fAllowMultipleFramesPerPacket(allowMultipleFramesPerPacket) {
    // Set up the "a=fmtp:" SDP line for this stream:
    char const *fmtpFmt =
            "a=fmtp:%d "
            "cpresent=0;config=%s\r\n";
    unsigned fmtpFmtSize = strlen(fmtpFmt)
                           + 3 /* max char len */
                           + strlen(fStreamMuxConfigString);
    char *fmtp = new char[fmtpFmtSize];
    sprintf(fmtp, fmtpFmt,
            rtpPayloadType(),
            fStreamMuxConfigString);
    fFmtpSDPLine = strDup(fmtp);
    delete[] fmtp;
}

MPEG4LATMAudioRTPSink::~MPEG4LATMAudioRTPSink() {
    delete[] fFmtpSDPLine;
    delete[] (char *) fStreamMuxConfigString;
}

MPEG4LATMAudioRTPSink *
MPEG4LATMAudioRTPSink::createNew(UsageEnvironment &env, Groupsock *RTPgs,
                                 u_int8_t rtpPayloadFormat,
                                 u_int32_t rtpTimestampFrequency,
                                 char const *streamMuxConfigString,
                                 unsigned numChannels,
                                 Boolean allowMultipleFramesPerPacket) {
    return new MPEG4LATMAudioRTPSink(env, RTPgs, rtpPayloadFormat,
                                     rtpTimestampFrequency, streamMuxConfigString,
                                     numChannels,
                                     allowMultipleFramesPerPacket);
}

Boolean MPEG4LATMAudioRTPSink
::frameCanAppearAfterPacketStart(unsigned char const * /*frameStart*/,
                                 unsigned /*numBytesInFrame*/) const {
    return fAllowMultipleFramesPerPacket;
}

void MPEG4LATMAudioRTPSink
::doSpecialFrameHandling(unsigned fragmentationOffset,
                         unsigned char *frameStart,
                         unsigned numBytesInFrame,
                         struct timeval framePresentationTime,
                         unsigned numRemainingBytes) {
    if (numRemainingBytes == 0) {
        // This packet contains the last (or only) fragment of the frame.
        // Set the RTP 'M' ('marker') bit:
        setMarkerBit();
    }

    // Important: Also call our base class's doSpecialFrameHandling(),
    // to set the packet's timestamp:
    MultiFramedRTPSink::doSpecialFrameHandling(fragmentationOffset,
                                               frameStart, numBytesInFrame,
                                               framePresentationTime,
                                               numRemainingBytes);
}

char const *MPEG4LATMAudioRTPSink::auxSDPLine() {
    return fFmtpSDPLine;
}
